Presently, there are numerous methods for reducing background noise in speech recordings made in adverse environments. One such method is to use two or more microphones on an audio device. These microphones are localized and allow the device to determine a difference between the microphone signals. For example, due to a space difference between the microphones, the difference in times of arrival of the signals from a speech source to the microphones may be utilized to localize the speech source. Once localized, the signals can be spatially filtered to suppress the noise originating from different directions.
Beamforming techniques utilizing a linear array of microphones may create an “acoustic beam” in a direction of the source, and thus can be used as spatial filters. This method, however, suffers from many disadvantages. First, it is necessary to identify the direction of the speech source. The time delay, however, is difficult to estimate due to such factors as reverberation which may create ambiguous or incorrect information. Second, the number of sensors needed to achieve adequate spatial filtering is generally large (e.g., more than two). Additionally, if the microphone array is used on a small device, such as a cellular phone, beamforming is more difficult at lower frequencies because the distance between the microphones of the array is small compared to the wavelength.
Spatial separation and directivity of the microphones provides not only arrival-time differences but also inter-microphone level differences (ILD) that can be more easily identified than time differences in some applications. Therefore, there is a need for a system and method for utilizing ILD for noise suppression and speech enhancement.